Method and apparatus for fast response and distortion measurement

ABSTRACT

A method and apparatus for fast response and distortion measurement of a signal transfer device. A computer processor generates a multitone test signal of predetermined duration and stores it in a memory. The test signal is read out, converted to analog form, if necessary, and applied to the input of a device under test. The output produced by the device under test in response to the test signal is acquired and digitized, if necessary, and a Fast Fourier Transform is performed on the acquired data to determine its spectral characteristics. Frequency response, harmonic distortion, intermodulation distortion, phase distortion, wow and flutter and other signal transfer characteristics are measured by the CPU by analysis of the output signal.

This is a continuation of application Ser. No. 07,764,142 filed Sep. 20,1991, now abandoned.

BACKGROUND OF THE INVENTION

This invention relates to methods and apparatuses for quickly measuringresponse and distortion characteristics of a signal transfer device, andin particular, for measuring the signal transfer characteristics of anaudio signal transfer device, such as an audio amplifier, a telephonechannel, or the like.

In the sound reproduction, broadcast and telephone communicationsindustries it is often necessary or desirable to determine the responseand distortion characteristics of a signal transfer device in order toevaluate, repair or improve an audio signal transfer path. The transferdevice may be any of many commonly known devices, such as a stereo soundreproduction amplifier (sometimes referred to as a "hi-fi" or "stereo"amplifier), a telephone communications channel or other communicationslink, a magnetic tape audio signal recorder, or audio signal broadcastequipment. What these devices have in common is that they accept as aninput one or more audio frequency signals and reproduce them at theiroutput, either immediately or delayed in time.

The conventional way of measuring response and distortioncharacteristics of such devices ordinarily has been to applysequentially one or more known test tones, i.e., audio frequencysinewave signals whose frequencies are known, to the input of an audiosignal transfer device and to measure their amplitude and relativephase, and the amplitude of their harmonics, at the output of thedevice. The tones are usually measured at the output using a bandpassfilter whose center frequency is tuned to one of the input frequenciesor a harmonic thereof. For example, to determine the frequency responseof such a device, a number of tones are applied sequentially to theinput and their amplitudes are measured at the output to obtain datarepresentative of output amplitude as a function of frequency.Similarly, to measure harmonic distortion a known frequency is appliedto the input of a device and the amplitudes of those harmonics of theinput frequency which are present at the output are measured as anindication of harmonic distortion. Conventional testing withmanually-operated equipment can take as much as an hour for a thoroughevaluation. This can be reduced with computer-controlled equipment, butstill requires a significant amount of time during which the deviceunder test (hereinafter "DUT") cannot be used for normal activity. It isto be understood that the terms device or DUT used herein refer withoutlimitation to one or more devices connected together.

A recognized international measurement standard has been adopted by theCCITT (International Consultative Committee on Telephone and Telegraph)and the EBU (the European Broadcasting Union) which employs twosequences of test tones, one for monophonic devices and one for stereodevices. The monophonic sequence lasts 31 seconds and the stereophonicsequence lasts 33 seconds. The sequence begins with a preamble whichindicates which sequence is being sent and the originator of the test.The tones are one second in duration each, and an 8 second long pause isincluded for noise measurements. However, while this method has theadvantage of being standardized, like other conventional test methods itis relatively time consuming; that is, it requires the device under testto be shut down from normal activity for a significant amount of time,thereby disrupting normal operations.

There are also devices known as real time analyzers that apply a whitenoise signal to the input of a DUT and provide an indication of theoutput amplitude at a number of frequencies by the simultaneous use ofmultiple bandpass filters tuned to those frequencies. The amplitudes ofthe outputs from those filters are displayed. However, those devices arelimited in their usefulness in that they provide little informationbeyond frequency response, and they require time to average the noisesignal at the various frequencies measured.

Thence, it can be seen that there has been a need for a method andapparatus for making thorough signal response and distortionmeasurements of signal transfer devices quickly, so as to take them outof service for a minimal amount of time.

SUMMARY OF THE INVENTION

The limitations of conventional response and distortion measurementmethods and apparatuses are overcome in the present invention by a formof digital parallel processing of the test signal input to a DUT and theresultant output. A plurality of selected test tones are applied to theinput of the DUT simultaneously. The output of the DUT is measured usinga Fast Fourier Transform ("FFT") to convert the time domain outputsignal to the frequency domain, thereby permitting the amplitude, phaseand frequencies of the output signal components to be determined.

After the type of test to be performed and the test tones are selected,the input test signal is created in digital form by a computer processorand stored in a digital memory. When the test is to be run, thedigitized test signal is read out of memory and converted to analog formfor input to the DUT, assuming that the DUT is not a digital audiosignal processor. The output from the DUT is simultaneously acquired,converted to digital form (assuming that the device does not have adigital output) and stored in the digital memory. In this way, the testsignal application and acquisition time can be reduced to the period ofjust a few repetitions of the test signal stored in memory. If the DUTis a digital audio signal processor, no digital to analog or analog todigital conversion is needed. A combination of analog and digitalequipment may need only one digital to analog or analog to digitalconversion.

Once the DUT output data is acquired, the DUT may be returned to normaloperation, and computation of the FFT is performed to identify theamplitudes, phases and frequencies of the DUT output signal. Values forsignal transfer characteristics such as frequency response, harmonicdistortion, intermodulation distortion, phase distortion, wow andflutter, and channel separation in a stereo or other multichannel systemcan be computed from test information displayed and produced innumerical form or as a function of some parameter, such as frequency. Inaddition, the source of the test signal can be identified by thepattern, or signature, of the DUT output signal.

Accordingly, it is a principal objective of the present invention toprovide a new and improved method and apparatus for measuring the signaltransfer characteristics of a signal transfer device.

It is another objective of the present invention to provide ameasurement method and apparatus that simultaneously applies a pluralityof test tones to the device to be tested and transforms the output ofthe device to the frequency domain to identify the amplitudes, phasesand frequencies of the components that make up the output signal.

It is a further objective of the present invention to provide ameasurement method and apparatus that digitally generates an inputsignal comprising a plurality of test tones simultaneously applied to adevice to be tested and converts the output of the device to digitalform for analysis.

It is yet another objective of the present invention to provide a methodand apparatus that permits the signal transfer characteristics of asignal transfer device to be determined in minimal time.

The foregoing and other objectives, features, and advantages of theinvention will be more readily understood upon consideration of thefollowing detailed description of the invention, taken in conjunctionwith the accompanying drawings.

BRIEF DESCRIPTION OF THE DRAWINGS

FIG. 1 is a block diagram of an apparatus according to the presentinvention.

FIG. 2 is a schematic diagram of the basic principle of operation of thepresent invention.

FIG. 3A is a plot of a typical sinusoidal waveform used in conventionalresponse and distortion measurements.

FIG. 3B is a plot of a typical DUT input waveform used in the presentinvention on the same relative amplitude scale as FIG. 3A.

FIG. 4 is an illustration of three tone signal waveforms relative to asignal generation time block during which an output signal is acquired,a first one having a period equal to that time block and two othershaving periods that are integer multiples of the frequency of the firstone.

FIG. 5 is an illustration of a plurality of frequency subdivisions, or"bins," corresponding to the allowable frequencies that can be measuredat the output of a DUT when using a Fast Fourier Transform according tothe present invention.

FIG. 6A is a plot of input test tone amplitudes as a function offrequency for a sixty tone test signal.

FIG. 6B is a plot of illustrative measured DUT output amplitudes as afunction of frequency for a sixty tone test signal.

FIG. 6C is a plot of input test tone amplitudes as a function offrequency for a sixty tone test signal, weighted to simulate programmaterial.

FIG. 7 is a plot of illustrative DUT output amplitudes as a function offrequency when excited with a single sinewave, showing harmonicdistortion components to be measured in accordance with the presentinvention.

FIG. 8 is a plot of illustrative measured DUT output amplitudes, atharmonic frequencies only, as a function of frequency for a five tonetest signal.

FIG. 9 is a plot of input test tone amplitudes as a function offrequency for a five tone test signal.

FIG. 10 is a plot of illustrative DUT output amplitudes as a function offrequency, simplified to show only two test tone intermodulation and wowand flutter components to be measured in accordance with the presentinvention.

FIG. 11 is a plot of illustrative measured DUT output amplitudes, atintermodulation frequencies only, as a function of frequency for a fivetone test signal.

FIG. 12 is a plot of an illustrative DUT input signal amplitude as afunction of time, for a sixty tone test signal with randomly selectedphase.

FIG. 13 is a plot of illustrative measured DUT output amplitudes, atselected unused frequencies, as a function of frequency for a five tonetest signal.

FIG. 14A is an illustration of a DUT output waveform whose period hasbeen lengthened by the DUT.

FIG. 14B is an illustration of window waveform.

FIG. 14C is an illustration of the product of the window waveform shownin FIG. 14B with the waveform shown in FIG. 14A, as found in accordancewith the present invention.

FIG. 15 is a plot of an illustrative DUT output amplitude as a functionof time for a multi-tone test signal, where one test tone has anamplitude about 12 dB higher than the amplitudes of the other testtones.

FIG. 16A is a plot of illustrative measured DUT output amplitude as afunction of frequency for a five tone test signal input, showingfrequency bands, not including the test tones, that are to be rootsquare summed.

FIG. 16B is a plot of the root square sum values computed for the bands(I, II, III, IV, V and VI) in ranges illustrated in FIG. 16A as afunction of the center frequency of the band.

FIG. 17A illustrates the identification of the presence of a pattern offrequency components in the output of a DUT having amplitudes exceedingpredetermined required component minimum amplitude thresholds.

FIG. 17B illustrates the identification of extra frequency components inthe output of a DUT having amplitudes exceeding a predetermined maximumthreshold.

DETAILED DESCRIPTION OF THE INVENTION

A preferred embodiment of the present invention and its relationshipwith a device under test ("DUT"), is shown in FIG. 1. An apparatusaccording to the present invention preferably comprises an input memory10 and an output memory 12, both of which preferably are random accessmemories which are part of a computer processor 14, and may besubdivisions of the same memory. The computer processor includes acentral processing unit 16 (hereinafter "CPU") which is interconnectedwith memories 10 and 12 and programmed to carry out the functions of thepresent invention.

Under control of the CPU 16, the input memory 10 supplies to a digitalto analog converter 18 stored data representing a digitized waveform tobe provided to the input 22 of the DUT 20. The waveform is predeterminedby the CPU in accordance with user specifications of amplitude, phaseand frequency of the test tones, and stored in the input memory 10 forquick access during testing of the DUT. The digital to analog converter18 converts the digitized waveform supplied by the input memory 10 to ananalog signal for application to the input 22 of the DUT. The output 24of the DUT is supplied to analog to digital converter 26 which digitizesthat analog signal and stores it in output memory 12. The results oftest computations made on that stored, digitized output data aredisplayed on a video monitor 28 or other suitable alphanumeric orgraphical display device. This enables the test data to be acquired inminimal time so that the DUT can be returned to service in a negligibleperiod of time, e.g., one second, after which the processing explainedhereafter can take place. At the same time, it is to be recognized thatthe method and apparatus of the present invention as described hereincould be implemented with other hardware and that the device under testcould be a digital audio signal processor which would not require thedigital to analog converter 18 or the analog to digital converter 26, orboth, and that it could have a digital input with an analog output or ananalog input with a digital output, without departing from theprinciples of this invention.

The present invention employs a type of parallel processing by applyingto the DUT 20 a plurality of sinusoidal test tones at once, asrepresented by f₁, f₂, f₃, and so on, up to f_(i), in FIG. 2, where i isthe number of distinct test tones applied to the device under test atonce. These tones are added, as represented by symbol 30, and applied tothe input 22 of the DUT 20. The output 24 of the DUT is measured andanalyzed to determine the spectral characteristics, that is, thefrequencies, phases and amplitudes of those frequencies which arecomponents of the output waveform of the DUT. Those frequencies, whichmay be fewer, but generally are greater than the number of inputfrequencies, are represented by w₁, w₂, w₃ and w_(o) in FIG. 2, where orepresents the number of distinct frequencies found in the outputwaveform. By analyzing the frequencies, phases and amplitudes of thecomponents that make up the output signal it can be determined whetherthere is harmonic distortion, intermodulation distortion, amplitudedistortion, or phase distortion. The amount of noise present, theexistence of wow and flutter, a change in speed of the signal as, forexample, where a tape recorder is running too slow or too fast, and inthe case of a multichannel system, the degree of separation of channels,can also be determined. The computer processor 14 may be programmed inany convenient way to perform the measurements set forth herein.

While the type of signal typically applied to the input of a DUT in theprior art ordinarily comprises one, or perhaps two, sinusoidal signalsas shown in FIG. 3A, the signal applied to the device under test in thepresent invention would ordinarily appear more or less like that shownin FIG. 3B, depending upon the specific frequencies, phases andamplitudes of the test signal frequencies f₁ through f_(i).

Turning now to FIG. 4, the period T represents the time during which theinput signal 22 to the DUT 20 completes one waveform repetition, and aninteger sub-multiple of the time during which the output signal 24 ofthe DUT 20 is acquired in the method and apparatus of the presentinvention. That is referred to hereinafter as a signal generation timeblock or "generation block." Waveform 32 represents a generation blockrepetition frequency having the period T. Waveforms 34 and 36 representtwo test tones, though there may be many more. The generation blockrepetition frequency is both the lowest frequency which can be generatedand the closest possible spacing of the test tones. Each of the testtones is an integral multiple of the generation block repetitionfrequency; for example, test tone 34 is two times the generation blockrepetition frequency of waveform 32, and test tone 36 is five times thegeneration block repetition frequency of waveform 32. However, it isdesirable that the test tones, and their harmonics, not be integralmultiples of one another, because that would make identification ofharmonic distortion at the output of the DUT more difficult.

The output measurement and analysis is accomplished preferably bysampling the DUT output in the time domain at a frequency at least twiceas high as the highest frequency component to be measured at the outputof the device under test, as dictated by the sampling theorem. Asindicated above, the sample data is digitized, stored and operated on bythe CPU 16. The preferred technique to be used is a Fast FourierTransform ("FFT") which converts the time domain data to the frequencydomain. However, other time domain to frequency domain transforms mightalso be used without departing from the principles of the invention. Inthe preferred embodiment of the invention, the DUT output signal issampled at the same rate as the test signal samples applied to theoutput. This causes all frequency components in the signals to fall inthe center of measurement FFT frequency bins. This technique is wellknown in the electrical engineering art, and many useful computeralgorithms can be found to implement the technique.

The output is measured over a period referred to hereinafter as ameasurement time block or "measurement block." The measurement block isan integer multiple of the generation block. The generation block periodT is typically chosen so that the measurement block has a number of datasamples equal to a power of 2, though this is not required for some FFTalgorithms. The FFT yields one distinct measurable frequency for eachpair of input samples. The minimum frequency resolution is equal to thesample rate divided by the number of distinct measurable frequencies.For example, for 16,384 input samples at a sample rate of 48,000 Hz,there will be 8,192 distinct measurable frequencies with a spacing of2.93 Hz. Preferably the lowest test tone for most audio systems would be20 Hz, or in the case of the aforementioned example, 17.58 Hz.

Since a fixed number of points i in the time domain are acquired foranalysis, the test tones may be generated digitally to exactly match themeasurement block. Also, the frequency spectrum available to be measuredat the output of the device under test may be conceptually divided intoa finite number of frequencies that can be separately identified,referred to hereinafter as "bins" as shown at 38 in FIG. 5. Each bin isseparated by the frequency equal to the highest resolution of thesystem. For example, if the block repetition frequency is 2.93 Hz, thecenter of each bin will be separated by that amount.

A commonly known characteristic of an audio transfer device is itsfrequency response. This is the variation of device gain with frequency.In the present invention, rather than applying a single frequency to theinput of the DUT and measuring its amplitude at the output and repeatingthe process for many different input frequencies, a plurality offrequencies are simultaneously applied to the input, and theircorresponding amplitudes at the output are identified by the FFT. In thefrequency domain the input would be as shown in FIG. 6A, where theamplitude is plotted as a function of frequency, on a log scale. In thisexemplary case, a 60 tone input is employed, though fewer or greaternumbers of input tones, or frequencies, may be employed. The amplitudeof the output is measured only at the input frequencies, and plotted asa function of frequency, as shown in FIG. 6B. In FIG. 6B the amplitudesof frequencies between those corresponding to the test tones have beeninterpolated in order to obtain a continuous curve. While interpolationis not necessary, it is desirable because it makes the output easier toread and estimation of amplitudes between the distinct test toneseasier.

In order to measure the frequency response of the DUT under nearlyactual operating conditions, the input amplitudes may be weighted as afunction of frequency to simulate frequency amplitudes in programmaterial, as shown in FIG. 6C.

Wow and flutter, a type of frequency modulation distortion, is oftenproduced by audio tape players. This has the effect of shifting energyfrom test tones into side bands of the test tones. The amplitudemeasurement of the acquired test tone frequency will produce anincorrect value; that is, it will be too low. This problem may be solvedby root sum square addition of the tone frequency and its neighboringsidebands.

Another characteristic of audio signal transfer devices is the extent ofharmonic distortion that they produce. In FIG. 7 amplitude is plotted asa function of frequency. An input signal having frequency f is shown,along with its second, third, fourth and highest identified harmonics,w₁, w₂, w₃ and w_(n). While this represents the output that would beobtained by a single test signal where the device under test producessubstantial harmonic distortion, a multitone test signal may be employedwith the present invention to obtain a comprehensive harmonic distortionmeasurement in a short period of time. For example, FIG. 8 shows a plotof output harmonic amplitude as a function of frequency where a fivetone test signal, as shown in FIG. 9, has been applied to the input ofthe DUT. The five test tones are selected to not be integer multiples ofeach other so that the harmonics of those tones will occur at uniqueplaces in the spectrum. A total harmonic distortion (hereinafter "THD")value can be obtained by taking the root sum square of the measuredharmonic distortion component output amplitudes; that is: ##EQU1## whereH=the number of harmonic components, and

A_(H) =the amplitude of a given harmonic component.

If test tones were chosen so that the harmonics were allowed to overlap,it would be possible for harmonics with opposing phases to cancel,thereby producing lowered, incorrect distortion readings. If the DUT hasa nonlinearity with a sharp discontinuity, it will generate high orderharmonics. Low frequency test signals will then have harmonics whichextend all across the audio band. This makes selection of test tonesdifficult if harmonic overlap is to be avoided. However, mostnonlinearities produce a distortion spectrum which falls off withincreasing harmonic order. This allows an upper limit on harmonic orderto be used to calculate which test tones will cause overlap. Above thatlimit the amplitude of harmonics are likely to be negligible whencompared with the components already included in the output. In FIG. 8the amplitudes between the various harmonic frequencies actuallymeasured are interpolated to provide a continuous curve.

Sometimes it is desirable to make a measurement of the harmonicdistortion performance of a device which can be compared to conventionalharmonic distortion measurements, yet the fast frequency response andseparation measurement capabilities of the present invention are stillneeded. Raising the amplitude of one test signal frequency componentsignificantly above the others will cause the waveshape and crest factorto be dominated by the larger tone. The harmonics created by the DUTwhen excited by this signal will be dominated by the harmonics of thelarger tone. In practice, an amplitude difference of about 12 dB issufficient to provide reasonable correlation with single sinewave basedmeasurements in most cases, although some have required a level ratio ashigh as 30 dB. Such a waveform is illustrated in FIG. 15. When thistechnique is used, the other test tones may still be used to obtain allother measurement capabilities of the present invention, such asfrequency response and channel separation measurements.

Measurement of intermodulation distortion (hereinafter "IMD"), and wowand flutter in the present invention is explained with respect to FIG.10. In FIG. 10 one frequency of a multitone input signal is shown byf_(b) and another, higher frequency test tone is represented by f_(a).Ordinarily there would be more than just those two test tones, but FIG.10 has been limited to those two tones for purpose of clarity. Thehigher IMD components are represented by (f_(a) +f_(b)), or "f_(a+b) ",and (f_(a) -2f_(b)), or "f_(a+2b) ". Similarly, the lowerintermodulation distortion components are represented by (f_(a) -f_(b)),or "f_(a-b) ", and (f_(a-2fb)), or "f_(a-2b) ". That is, intermodulationbetween f_(b) and f_(a) produces sum and difference frequencies in theoutput. It is recognized that more distortion products between those twotones may be generated , but only these are shown for the purpose ofexplanation. Wow and flutter, a type of frequency modulation distortion,is often produced by audio tape players. That also produces distortioncomponents, shown in FIG. 10 as (f_(a) +f_(c)), or "f_(a+c) "; (f_(a)-f_(c)), or "f_(a-c) "; or (f_(a+d)), or "(f_(a) -f_(d)) where f_(c) andf_(d) represent wow and flutter frequency components.

If the DUT has nonlinearities, there will be IMD components between allcombinations of tone frequencies. By modeling the nonlinearity as apower series, the IMD frequencies can be predicted. These IMD componentswill appear at frequencies above, below and between the test tones. Thecalculation of these IMD frequencies becomes very complex when many testtones are included and when the nonlinearity is of a high order. Thence,selection of the test tones must be done carefully.

In applications involving large numbers of test tones, i.e., about 6 ormore, the distortion spectrum will become complex enough that althoughsome components may cancel there will be a sufficient number ofdistortion components remaining that an accurate assessment ofnonlinearity may still be made. However, the sheer number of distortioncomponents involved may make the graphing time excessive or theinterpretation difficult. By adding all components except the originaltest tones themselves, a measurement is made which does not suffer fromthese limitations.

To allow correct summation, a root-sum-square (often called root squaresummation) technique is used to produce an indication of the effectivelevel in the distortion components. If the measurements are performedusing the results of an FFT this merely involves root square summationof the bin levels for all bins excluding those of the original tones.The fact that frequencies are included which may not have distortioncomponents will cause the effects of noise to be included in themeasurement. This makes the measurement analogous to the total harmonicdistortion plus noise (THD+N) measurement which is commonly performedwhen using a single sinewave stimulus. Since a multiple sinewavestimulus is used, the measurement will also include intermodulationproducts and is most accurately called total distortion and noise. Ifsome indication of the distortion variation with frequency is desired,the summation may be done in parts, or frequency bands, as illustratedin FIG. 16A, each band covering a portion of the frequency range, andthe sums graphed as a function of the center frequency of that band, asshown in FIG. 16B.

In cases where wow and flutter is not a significant error source in themeasurement, a numerical representation of IMD may be obtained by rootsquare sum addition of the components at the intermodulationfrequencies. A visual display of IMD is obtained by plotting theamplitudes of the output at IMD frequencies only, as a function offrequency, as shown in FIG. 11. The amplitudes of frequencies betweenthe IMD components are interpolated to provide a continuous curve.

In order to obtain a numerical value representative of IMD when wow andflutter is present, bins located symmetrically on either side of a testtone are summed linearly in pairs, but with a phase inversion of one ofthe frequency components in each pair. Then the pairs are root sumsquared; that is: ##EQU2## where N=the number of sideband pairs, and

A_(UN) =the upper sideband, and

A_(LN) =the lower sideband.

The inner subtraction is of the vector amplitudes which include phaseinformation. This produces a value for IMD sidebands, but rejects wowand flutter sidebands in the spectrum measured. This is because theupper and lower wow and flutter sidebands are in phase (so they cancelout when one is inverted), while the upper and lower IMD sidebands are180° apart in phase (so they add when one is inverted).

To obtain a wow and flutter distortion numerical value, the analysisbins located symmetrically about a test tone are summed linearly inpairs (without inversion of one). Then the pairs are root square summed,thereby canceling out the IMD sidebands in the spectrum measured; thatis: ##EQU3## where N=the number of sideband pairs, and

A_(UN) =the upper sideband, and

A_(LN) =the lower sideband.

The inner addition is of the vector amplitudes which include phaseinformation.

The distortion measurements obtained by this technique are not directlycomparable to those obtained by single sinewave total harmonicdistortion plus noise ("THD+N") testing, or by conventional IMD testing.This is because the crest factor, i.e., the ratio of peak to RMSamplitude as illustrated in FIGS. 3A and 3B, of a multitone inputsignal, will always be higher than that of a single sinewave; that is,for the same peak signal amplitude, the amplitude of each individualtone will be lower than a single sinewave at that tone. However, anaccurate and reliable result may be obtained by this technique toprovide good comparison of DUT characteristics over time, or to otherdevices tested by the same technique. It may be desirable to scaledistortion measurements by a correction factor to yield readings whichare comparable to that obtained with conventional testing for typicalDUTs. This can be done by measuring similar DUTs in the conventional wayand deriving correction factors by comparing the two sets ofmeasurements.

In order to minimize the crest factor, and thereby reduce the likelihoodof clipping distortion, the test tone signals may be provided withrandom phases, as shown in FIG. 12 for a sixty tone test signal.Alternatively, the crest factor can be made similar to that of programmaterial by fixing the phase of each of the sinewave components of theinput signal to create a test signal with the desired crest factor. Thisis because, since all the test tones are a multiple of the generationblock repetition frequency, their phases are locked together.

Noise produced by the DUT may be measured by selecting test tones whichleave gaps in the harmonic and intermodulation distortion componentspectrum, and measuring the amplitude of the output signal at frequencybins between the test tones and their harmonic and IMD frequencies. Aplot of the amplitude values obtained by this technique, as a functionof frequency, with interpolation between distinct frequencies, is shownin FIG. 13. A numerical noise value measurement can be obtained by rootsquare summing the amplitudes obtained this way. However, the squaredand summed value must be multiplied by a constant representing thenumber of bins used in the computation, the bandwidth of the bins, andthe bandwidth of the measurement to yield an accurate wide band noisefigure. The computation is made as follows: ##EQU4## where N=number ofbins used in noise calculation,

A_(I) =amplitude in bin number N,

N_(B) =number of bins across the entire output measurement spectrum; and

C_(WG) =window gain correction constant, as commonly known in the use ofFFTs.

If enough frequency points are used it is possible to factor inweighting filter gain as a function of frequency when computing noise,thereby yielding a weighted noise measurement. This is done bymultiplying each A_(I) by the weighting filter gain at each binfrequency before the squaring and summation in the above equation.

When using signals with a large number of test frequencies it may bedifficult to find empty bin frequencies. By making the measurement blocklength an integer multiple larger than one of the generation blocklength, there will be additional resolution in the measurement spectrum.This will ensure empty bin frequencies since the generated frequencies,and the resulting harmonic and intermodulation frequencies, must alwaysbe an integer multiple of the generation block repetition rate. Bymaking the measurement block length twice that of the generation blocklength, every alternate frequency bin will be guaranteed to be free ofdistortion products. Consequently, summing only the alternate bins willresult in a noise measurement uncorrupted by distortion.

Applying different test signals to two channels of a stereo DUT allowsmeasurement of channel separation. The two signals may be very nearlyidentical, but with each missing a few test tones that the othercontains across the audio spectrum employed. By measuring the amplitudeof those missing tones in the output of the DUT, crosstalk from theopposite channel may be determined. This also allows a quick check foraccidental reversal of the channels.

FFT phase values will reflect the phase of each component in the DUToutput signal relative to the start of the acquired data. If absolutephase information is desired, it is easily obtained from a singlechannel of data. The phase of each component in a multitone test signalis known since the waveform is software created and digitally generated.If the test tones have been specified to all be in phase, the measuredphase will be the phase shift through the DUT. When measuring audiosignal transfer devices that introduce time delay, the delay will bereflected in the measurements. Since the added phase shift will beproportional to the product of time delay and frequency, it isstraightforward to compensate for it, if necessary. In situations wherethe signal generation means and the signal measurement means areco-located, it is possible for the generation means to trigger theacquisition means with the generation of a new time block so as toprovide an absolute time reference and enable phase measurements throughthe device under test without knowing the phase relationship of theoriginal tones. The channel-to-channel phase of a stereo system may beobtained by subtracting the phase data of one channel from the phasedata of the other.

Any of the preceding performance measures computed from the spectralcharacteristics of the DUT output signal, such as harmonic distortion,intermodulation distortion, phase response, etc., may be compared topredetermined maximum or minimum values, or limits, based uponpredetermined criteria. This allows "pass/fail" testing of devices to beperformed using the fast measurement method described herein.

Since the FFT requires an integral number of cycles during themeasurement block, a slowdown or speedup of the test tone signal at theoutput requires compensation. This can be accomplished in either of twoways.

In FIG. 14A, one frequency of a test signal that has been slowed down bythe DUT is shown in relation to the measurement block where themeasurement block is equal to the generation block. It can be seen thatby restricting the data points to those during the period T adiscontinuity exists at point 38. This is recognized by the CPU when theresults of the FFT do not fall in the center of a bin. They may evenfall in another bin in this situation. However, the data acquired can becompensated by interpolation or by the use of a window.

Where interpolation is to be used, the CPU measures the frequencies inthe output signal from the DUT, compares one or more of thosefrequencies to the frequencies present in the input signal to the DUT,computes a percentage by which the signal is too slow or too fast, andinterpolates between the pairs of sampled data by that percentage to getcorrect values. The FFT is performed again. For example, assume that t₁and t₂ represent adjacent samples of the instantaneous amplitudes of thewaveform in FIG. 14A. Those amplitudes will each be incorrect for asignal assumed to have an integral number of cycles during period T. Thecorrect value for t₁, that is, the value if the signal had not beenslowed down, can be computed by interpolating between the amplitudes att₁ and t₂ by the appropriate percentage of time and then assigning thatamplitude to t₁ for purposes of the FFT. This would be repeated for allof the samples during the time period to reconstruct a continuouswaveform.

Alternatively, the waveform in FIG. 14A may be multiplied by a windowwaveform, such as that shown in FIG. 14B. The window waveform has apredetermined shape whereby it starts at zero amplitude at the beginningof period T and ends at zero amplitude at the end of period T. Thatensures that the product will also start at zero and end at zero, asshown in FIG. 14C, thereby eliminating the discontinuity. Thereafter,the FFT may be applied to the block of data and the spectral informationused as before. A tradeoff in the use of this second technique is thatthe frequency resolution of the FFT is reduced by the window and thiswill limit the ability to use closely spaced test tones.

The specific test tones which are to be used may be user selected fromthose that are allowable (integral multiples of the block repetitionfrequency, as explained above). Since the block repetition frequencypreferably is not an integral number itself, the allowable frequencies,e.g., 17.58 Hz, etc., are unlikely to be intuitive choices of the user.Consequently, the frequencies actually chosen by the user are rounded ortruncated to the nearest allowable test tones. For example, 20 Hz wouldbe rounded to 17.50 Hz and 100 Hz would be rounded to 99.62 Hz.

Signals sent from one location to another often need to be identified asto their point of origin. In broadcasting there may be a need to knowwhich of four studios in separate cities is sending a test tone to thereceiving transmitter site. In the present invention this identificationmay be provided by defining a unique combination of test frequencies foreach source. Ordinarily it is sufficient to use identical signals foreach studio with the addition or deletion of one frequency uniquely foreach source. It is also possible, and usually preferable, to keep theamplitude spectra the same and alter the phase of one of the tones inthe signal. If there are tones close together in frequency the phaseshift of the DUT, e.g., a communications channel from one city toanother, may be quantified by these tones and the phase of theinformation carrying tone is easily recovered.

It is desirable for the signal generation equipment to be able to insertthe test signal into a normally operating audio transmission link andhave the measurement equipment automatically detect the presence of thetest signal and perform the measurement. To enable this automaticoperation, a reliable method of triggering is required which preferablydoes not add appreciable time to the measurement. Prior art automaticmeasurement systems have employed a frequency shift keyed ("FSK") tonesequence immediately preceding the test signal to trigger the start of ameasurement procedure at the receiving end. In the present invention itis possible for the measurement equipment to sense the presence of thetest signal by comparing the received spectrum to a pair of templateswhose shapes are determined by the test signal spectrum. This isillustrated in FIGS. 17A and 17B.

The technique requires a minimum percentage of originally generated testsignal frequencies to be present in the output signal from the DUT. Itis undesirable to require all of the tones to be present since the DUTmay have filtered some of them out due to its limited bandwidth or itsparticular frequency response. FIG. 17A shows a "template" 40 overlayedon a plot of the output spectrum of the DUT to identify tones thatshould be present. For example, if this minimum percentage had been setat 75%, then 25% of the 6 original test tone frequencies used in FIG.17A, or 4 original tone frequencies, must exceed the required componentamplitude threshold. In actuality, the template is a representation ofthe digital signal analysis process of checking the FFT data forselected frequency components.

A second part of the technique is to determine how many extra frequencycomponents are present and reject the signal as a test signal if toomany extra components are present. An extra frequency component isconsidered to be present if it exceeds a predetemined amplitudethreshold. This allows a few interfering tones as might be present fromac power line hum or video sync buzz to be present in the receivedsignal.

FIG. 17B shows a template 42 overlayed on the plot of the outputspectrum of the DUT to identify frequency or tone components that exceedan amplitude threshold. This threshold would typically be a percentageof the largest component amplitude, for example 1% or -40 dB. A maximumnumber of such tones would be allowed. The presence of more than thisnumber of tones whose amplitudes exceed the threshold would disqualifythe signal. The number might be fixed, for example at 5, or it might bea percentage of the number of original tones, for example 50%. For theexample of 6 original tones, a 50% limit would reject any signal whichhad more than 3 tones whose amplitude exceed the amplitude threshold andwhose frequencies were not present in the orignal signal. It is unlikelythat program material would have a sufficient number of the requiredfrequencies present and simultaneously have less than the allowableamplitudes of tones to pass both template tests and be mistaken for testsignal.

The measurement techniques described thus far herein have requiredknowledge of the original test signal frequencies. One way to accomplishthis is for the values to be programmed into the measurement system whenthese values are used to specify or create the test signal. However, ifthe test signal is stored as sample values in memory for later use thesefrequency values are not necessarily known. This would be the case for atest signal which was recorded on tape for playback testing of anothertape recorder or for a signal received over a communications link.Consequently, this invention includes two solutions to this problem.

If the waveform exists in memory as sample values but the originalfrequency values are no longer available, the original frequency may beobtained for the purposes of the measurements by Fast FourierTransforming the signal to be generated and identifying all frequencycomponents in the transformed data. This will then be a complete list ofthe test signal frequencies and will be identical to the original listused to generate the waveform samples.

If the signal is received and the original generated signal is not inmemory, as might occur over a communications link or from a test tape,an acceptable estimate of the original frequency list may be obtained bythe following method. After an FFT is performed on the acquired signalthe largest amplitude frequency component is determined. A prudent rangeof largest to smallest amplitude for the original tones is assumed. Thefrequency spectrum is searched for all frequencies which contain morethan this minimum amplitude signal. These frequencies are then assumedto be the original tones. The range must not be assumed to be too largeor the distortion products in the received signal may be mistaken fortest tones. If it is assumed to be too small some original test tonesmay be missed due to frequency response variations in the DUT. Inpractice a range of 30 dB is a reasonable compromise between theseconflicting requirements.

It is to be recognized that the computations and analyses describedherein preferably are made by the CPU 16 in accordance with theinstructions of a digital computer program. The CPU may be programmed avariety of different ways, as would be straightforward to a person ofordinary skill in the art.

The terms and expressions which have been employed in the foregoingspecification are used therein as terms of description and not oflimitation, and there is no intention in the use of such terms andexpressions of excluding equivalents of the features shown and describedor portions thereof, it being recognized that the scope of the inventionis defined and limited only by the claims which follow.

What is claimed is:
 1. Method for measuring the signal transfercharacteristics of a signal transfer device under test, said devicehaving an input and an output, comprising the steps of:(a)simultaneously applying to said input of said device under test aplurality of selected, distinct, substantially sinusoidal test toneswithin a selected bandwidth, the test tones having frequencies that areinteger multiples of a predetermined frequency, at least one of saidtest tones having a frequency that is a noninteger multiple of thefrequencies of the other said test tones; (b) acquiring a signal fromsaid output of said device under test; (c) performing atime-to-frequency transform on said acquired signal to obtain thespectral characteristics thereof; (d) detecting simultaneously all ofthe frequency components, and amplitudes thereof, of said acquiredsignal across said bandwidth, exclusive of the frequencies of said testtones; and (e) combining amplitudes of detected frequency components ofsaid acquired signal to obtain a measurement of non-linear effects insaid device under test.
 2. The method of claim 1 wherein step (e)comprises root square summing said amplitude of detected frequencycomponents in said acquired signal to compute a measurement of totaldistortion plus noise.
 3. An apparatus for measuring the signal transfercharacteristics of a signal transfer device under test, said devicehaving an input and an output, comprising:(a) application means forsimultaneously applying to said input of said device under test aplurality of selected, distinct, substantially sinusoidal test toneswithin a selected bandwidth, the test tones having frequencies that areinteger multiple of a predetermined frequency, at least one of said testtones having a frequency that is a noninteger multiple of thefrequencies of the other said test tones; (b) acquisition means foracquiring a signal from said output of said device under test; (c)transform means for performing a time-to-frequency transform on saidacquired signal to obtain the spectral characteristics thereof; (d)detection means for simultaneously detecting all of the frequencycomponents, and amplitudes thereof, of said acquired signal across saidbandwidth, exclusive of the frequencies of said test tones; and (e)combining means for combining amplitudes of detected frequencycomponents of said acquired signal to obtain a measurement of non-lineareffects in said device under test.
 4. The apparatus of claim 3, whereinsaid combining means comprises means for root square summing a pluralityof frequency components across a frequency band covering a portion ofsaid bandwidth of said acquired signal to obtain a measurement of totaldistortion plus noise associated with said frequency band.